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If one has a 2-D array and would like to take the 1-D Fourier transform along a direction $\theta$ degrees off the horizontal, is there a better/faster way to do this rather than rotating the image by $-\theta$ degrees, taking the 1-D FT across the columns, and then rotating the result back by $\theta$ degrees?
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Recently I read about image noise. I read about noise characteristics, but I am unable to understand the following two points:
The sources of noise in digital images arise during image acquisition
(digitization) and transmission.
Imaging sensors can be affected by ambient conditions
Interference can be added to an ima... |
I have a time series of data with columns labeled Months and Rows labeled Years. So, my data looks as:
1900 -5 8 8 6 15 ......
1901 -3 8 8 6 15 ......
1902 -12 4 8 6 15 ......
1903 -34 1 8 6 15 ......
1904 -5 8 8 6 15 ......
The ab... |
what kind of filtering is it? How signal/noise ratio in the output mix of the duration of accumulation would look like, knowing
Signal/noise ratio=0.1
,The number of cycles of accumulation of up to 200
,Limits of change in the signal/noise ratio of 0.1-3
|
Since Fourier transforms are applicable to only periodic functions, and unit impulse function{..0,0,1,0,0..} doesn't seem periodic, is this possible?
Is it like we are considering unit impulse function to be periodic in the range $(-\infty,+\infty)$?
|
I'm surveying for feasibility techniques for a potential future project that's rather out of my past areas of expertise, and I have a fairly basic question. Of course if it looks feasible I will have to learn a whole lot more on the topic.
What is the equation to compute the number of FFTs/IFFTs per sample required, gi... |
I'm currently working on a project that involves in Speech and recognising Phonemes (in kind of real-time).
The current algorithm works by always listening for input from the Mic, and input that is above a specific threshold then this is recorded to a .wav file and sent to the server for processing. E.g.
listening...
... |
I am trying to understand the MACE correlation filter as per the paper
Mahalanobis, A., Kumar, B.V.K.V., Casasent, D.: Minimum average correlation
energy filters. Appl. Opt. 26, 3630–3633 (1987)
The equation to train the MACE filter is this
I however am abit confused about what values you use for u. From my readin... |
Investigation of the method of suppression of random noise by coherent signal accumulation
Purpose - to identify opportunities for the coherent accumulation for cases of stationary and quasi-stationary signal.
Suppose that the input is the mixture of observed signal and a random white noise (i.e. noise with uniform spe... |
Goal: I have a video stream from a webcam. I need get frequency of the green color changes on the video.
Video color space is RGB
The signal looks sinusoidal with a frequency range 0.5 - 2.4 Hz
I performed the following steps:
Capture Video frames into collection.
Processing frames by Gaussian pyramid down level 4. (f... |
Hi everybody. I'm very new in Kalman filter. Today I try to design Kalman filter to get estimated postion, velocity and acceleration from measurement position (by linear encoder). I don't use optimal input $u_k$ and my Kalman equations are:
$\displaystyle \quad\
\hat x_k^- = A\hat x_{k-1}
$
$\displaystyle \quad\
P_k^... |
I have to compare two .wav files.
I am converting the wav file to an MFCC file. Why am I getting different MFCC matrices when using different Matlab toolboxes (one of them is giving a matrix having 20 rows while the other two are giving matrices having 13 rows, and even those two matrices are not same)?
Why are the t... |
I need to append (not add) a bandlimited signal F(b) to a bandlimited signal F(a) and keep the result bandlimited without modifying the part that corresponds to F(a).
Both are bandlimited to the same frequency (for example to 1/4 the Nyquist frequency). If F(a) is 100 samples long and F(b) is 100 samples long, the resu... |
I have a current signal represented by an array in python, which I would like to add shot noise to. Shot noise is defined as
$$
σ_i = \sqrt{2qI \Delta f}
$$
where $q$ is the elementary charge, $I$ the current and $\Delta f$ is the bandwidth of interest. I guess the simplest way to add the noise would be to to add it ... |
So I am getting to grasps with Fourier transforms. Intuitively now I definately understand what it does and will soon follow some classes on the mathematics (so the actual subject). But then I go on reading about the laplace transform and there I kind of lose it. What is the moment of a signal? Why is the fourier trans... |
How can we determine whether the receive filter is optimal or not in a digital receiver?
For a decision region, how can we choose a noise value zk?
|
Could it bring any meaning? Is this kind of convolution be useful is solving anything?
|
I have an ADC that will have a guitar track as an input. I want to find which notes are hitting via the frequency domain. Given that guitar frequency range is from 82Hz to 1046Hz which sample rate (window) should I choose? I had read on an article about some kind of rule but I can't find it..
|
I am trying to implement an EKF to estimate my position and velocity states by using accelerometer measurements as well as periodic GPS (position) measurements. Basically I want to use the constantly use the accelerometer and re-calibrate it for every position measurement that is available. When there is GPS dropouts... |
I delved into the state of the art of algorithms for Poisson noise estimation in order to estimate the variance, I found that the Expectation Maximization algorithm is very used and it is very effective and easy to use,
But I have not found the matlab code, I found only one that is was used for classification,
can you ... |
I'm a new guy in image processing and computer vision, so this question might be stupid to you.
I just learned some feature detection and description algorithms, such as Harris, Hessian, SIFT, SURF, they process images to find out those keypoints and then compute a descriptor for each, the descriptor will be used for ... |
First off: I'm really new in the signal processing department and MatLab as well, so please bear with me here.
What I want to achieve is to split my audio signal into smaller 25ms frames, where the frames overlap with 50%, apply a window function, thus the overlapping (probably Hamming or Hann), and then add it togethe... |
i am looking for discrimination of speech from animal sound if you take a 150ms frame of speech and bird as shown below, what feature can i extract from them so that i can effectively state the difference
zero crossings ?
cross correlation of successive frames seems high in bird sound ? isn't it
any other feature fr... |
Is there a function in MatLab to scale a given signal to a specific range, like how soundsc does it with the input?
I have to signals, one is a wave file, using wavreader (I know one should use audioreader now) to load the file, and another file that is being read using fopen and fread.
Now I want to compare those to s... |
Image registration algorithms can be classified into intensity-based and feature-based algorithms, I want to know When and Why should i choose intensity or feature based algorithm?
another question Is there a difference between image registration and optical flow?
|
Suppose the input signal is represented as $$y=x+w$$ where $x$ is the signal to be estimated and $w$ is AWGN. I design a high pass filter $H$. What does $$H^{T}Hw$$ represent? Will it be just AWGN but attenuated or will it AWGN completely distorted. We can also assume that $\sigma_w = 0.2$
|
I am looking for -or trying to create- a filter with a piece-wise monotonic second derivative in such a way that when placed on a non-periodic input signal, changes in sign of the second derivative happen as soon as possible and piece-wise monotonicity of the second derivative (so also of the first derivative and the f... |
In various software defined radios, there are three important parameters to set when receiving: frequency, bandwidth and MSPS (million samples per second?). What does it mean to receive with a triple of parameters $(freq, b, s)$ (freq-frequency, b-bandwidth, s-MSPS)? Radio listens to frequencies in $[f-b/2,f+b/2]$ rang... |
I want to program a microcontroller in C but I am asking about the theoretical aspect of it. I think I've got an idea what I am going to do but there is one thing I can't quite get. To begin with, I will use a guitar, so there is a 82Hz-987Hz range. So roughly, I'm going to low-pass the data on 1KHz, "window" them and ... |
I am making preparations to implement FFT on a wav File.wav Only Java.
I'm reluctant to use matlab
I have a File wav stereo, 16 bits.
I need to calculate the FFT, but I have confusing.
My File has a Frequency of 44100, the samples in bytes are 10 minuts ->
(10 minuts)*(60 seconds)*(44100 sample rate)*(2 bytes because ... |
I am looking at working broadly in the area of Speech Signal Processing, and would be grateful if someone can clarify these doubts!
1) Are there any differences between Processing and Analysis of Speech Signals, in terms of the scope they provide for research? Arent both of them concerned in detecting the spoken langua... |
I am reading about different integral transforms. Fourier transform being a subject I will soon take a class on. Looking at the wavelet transform there is one term that I do not understand, that is the noiselet. What would a noiselet look like and how is it defined? Is it some basis of the wavelet transform? It is hard... |
What is the minimum period P (in samples) of the signal $e^{j(\frac{M}{N} )*2πn}$ for the following values of M and N?
M=1,N=3
M=5,N=7
M=35,N=15
I have got the answer for the first pair of values, which is 3. But I could not get the other two correct. Please help.
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In a soundfile, I'd like to remove a frequency and all its harmonics. What is the best solution in your opinion ?
Do a loop with 20 notch filters, one after another.
// f is the fundamental frequency
for (i=0; i++; i<21)
{
file.notchfilter(f*i);
}
Question : does each notch filter create some unwanted side-effects, ... |
I was reading this paper for a project work.
We train on 1.6 million 32*32 color images that have been preprocessed
by subtracting from each pixel its mean value over all images and then
dividing by the standard deviation of all pixels over all images.
I've trouble distinguishing between "from each pixel its mean... |
I have got this question: let's suppose that I have three signals, representative of the same variable, but taken on three different component of a same device; in my case, three load signals, representative of the out-of-plane bending moment Mx of each of the blade of a wind turbine.
Is it any possible to effectively ... |
I have the following inverse system
$$G(s)= s^2 + 2s + 3$$
How do I implement it in Simulink? Note that the transfer function is only accepted if and only if the order of the numerator is less than the order of denominator.
|
I do not seem to know how to extract from a Matlab plot. For example in the plot given below, I have a simulated $P_b$ for Rayleigh fading and theoretical AWGN channel Pb plot against SNR ($\large \frac{E_b}{N_o}$).
My question is how do I get the value of $\large \frac{E_b}{N_o}$ for a particular value of $P_b$. For ... |
I am trying to recreate the following results:
...from the following data: Dropbox Link (FS40.dat)
The time is in milliseconds (frequency is 40000 Hz)
The article state that they used Complex Morlet wavelet to create the spectrogram:
"Power estimates from the averaged LFPs were calculated from
time–frequency spectrogr... |
I am learning Fourier analysis and without any teacher, just trying to read books on my own. I think I have made some decent progress but they are a couple of points which are still very unclear to me and that I can't find explained in any of the books I have.
One of the sources that I found in this document, which is ... |
Is it possible to detect if a sample of human voice by smartphone/computer(44Khz) is sourced from another electrical device or originated from a human being
I want to build a model in matlab such that, samples a vocal signal and decides whether it is a real human being talks a live or recorded, is it even possible?
|
I need some information about Stockwell transform (also known as the S-transform):
How can I implement it in MATLAB?
*Does it give you the damping ratio $\zeta$ of a signal like the complex Morlet continuous wavelet? $$x(t)= e^{-\omega_{n}\zeta}\sin(\omega_{d}t+\theta )$$
What are its advantages compared to CWT?
The ... |
How to find bandwidth of a composite signal that can be decomposed into five sine waves with given frequencies and having all peak amplitudes same ?
|
I am involved in designing a 10th order high pass filter with cutoff frequency 300Hz.
My hardware AIC3204 has a 5 biquads of direct form I, 24bit coefficient
so I have chosen the FDA (filter design and anylysis) tool of MATLAB to design a 10th order filter and convert it to SOS sections 5 such and realize coefficients ... |
I have a signal. I applied Gaussian smoothing to it and then for baseline reduction I appied Tophat filter to the smoothed version.
This is the original signal:
This is the final signal:
I read that KL Divergence helps in finding the information loss between two signals but then again a condition was that the element... |
Reading up on causality, I understand the mathematical definition, in so far as that a causal system, is one where the output depends only on the current time, and possibly the past time, but never the future times.
I have also seen that we can 'make a (transfer) system causal' by 'shifting it in time by an appropriat... |
I need to explore the algorithm of the digital guitar tuner or the digital guitar effects algorithms.
Is there any Book or articles that you recommend of?
All I know is that relate to DSP.
thanks
|
Black rectangles on these charts show same signal, which is actually color sequence from a image, taken from different positions.
It is easy to see that all images contain same pattern. Is here some algorithm which can detect these similarities, ideally implemented in Python ?
What about cut signals in multiple win... |
Let's say I have an IIR filter that is a notch type with a peak at -20db. How do I modify the coefficients of the filter so that the peak is at -10 db for example?
I know I can scale the denominator coefficients so that the global peak amplitude can be set, but it affects the whole filter, which I don't want.
Here is ... |
The Wikipedia page on sound localisation states:
Localization accuracy is 1 degree for sources in front of the listener
and 15 degrees for sources to the sides. Humans can discern interaural
time differences of 10 microseconds or less.
However, I can't find any more detailed information on localisation accuracy.
... |
I'm an undergraduate student and I am currently trying to find suitable topic for my bachelor thesis. The broader topic I've selected is "Image processing on GPU", so basically I'll be developing image processing algorithms specifically for GPU.
The problem is, I need to do a novel approach, not reinvent a wheel. Ther... |
So, this is a picture of saccadic eye movement reflected by an EOG. How can I determine if the eye movement system is linear, causal, memoryless, and time invariant from this displayed waveform? Please explain.
|
I have position $\theta(s) = \frac{s^2+1}{s^4+2s^3+3s^2+4s+5}$ and velocity $\dot\theta(s) = \frac{s(s^2+1)}{s^4+2s^3+3s^2+4s+5}$.
How do I construct 2-state-observer with sampling time $T_s$?
$\left[
\begin{array}{c}
\theta_{k+1}\\
\dot\theta_{k+1}\\
\end{array}
\right]
=A\left[
\begin{array}{c}
\theta_{k}\\
... |
What do you think of this notch / band reject filter for a very very narrow band ?
Original signal x[n]
--> FFT
--> then I just put "0" in the FFT bin of the frequency I need to reject (notch filter)
--> inverse FFT
--> output signal y[n].
How is called such a filter ? Does it really work, or is it too simple to be ... |
I have an application that I must use a complex Morlet wavelet function (cmorfb-fc), I can run CWT (continuous wavelet transform) and it's fine, but I want to run DWT(Discrete wavelet transform) with this complex Morlet(cmorfb-fc) function too. then I want to compare the DWT and CWT results.
so Does anyone can help and... |
What's the simplest way to implement a FIR filter in C/C++ ?
(I don't want to have to use a third-party software like Matlab or Octave in order to compute the coefficiens because I have to use different filters on thousand of files, with different frequencies for each file.)
Thus, is there a ready-to-use C/C++ DSP libr... |
So I am Designing a system which would be able to send and receive stereo voice signals.( like a telephone but with full-band stereo voice signals)
here is an example to make my question clearer.
In a music recording studio two microphones are foreseen, one for singer's voice and the other for the guitar's sound. The ... |
I have a digital on/off signal sampled at a regular interval. There was a problem with the sensor that produced the signal which has sometimes caused a regular on/off ticking when the sensor should be in the off state. I.e. sometimes when the readings should be zero, there is a square wave instead (but with an unequa... |
How edges of the triangle get on both sides of fs? What is their purpose? Why fs in the center? (from Lyons Understanding DSP)
|
is that possible to find SNR of an environment ? ie., i have a outdoor environment where i cannot find the clean speech ,if so i want to measure the signal power to noise power how to do that
can i do some thing like put a voice activity detection and find voice frames and find the voice frames power to non voice frame... |
I've got an impulse response of a hall, it's 10 seconds long.
Now say I want to apply this reverb to a wav file of some singing. I would perform a convolution of the data in two wav files and that the result would be a third lot of data, this time with the reverb. Great.
But how would this actually work in a real time ... |
This is a photo of our homeworld, taken by the Juno spacecraft recently, sling-shotting on its way to Jupiter. What it gained in speed, we lost in ours, however thankfully we will not be falling into the sun.
I think the South American continent is on the left.
However, we can notice that there is a sort of artifact... |
I am doing a personal project which aims to detect infant cry. I have an infant cry audio database and extracted the MFCC from these files using MATLAB. I've come up with 12 coefficients for each frame the audio was divided into.
My goal is to compare audio obtained from a microphone with the training data (MFCC from t... |
I am using impulse responses from http://sound.media.mit.edu/resources/KEMAR.html
These are available as either raw or diffuse-field equalized HRTFs.
My question is: Which do I want? Raw, or diffuse-field equalized?
I am using these IRs to project single notes (generated synthetically using a ring resonator) to 12 loca... |
I am experimenting with HRTFs.
Specifically I'm trying to create 12 impulse responses that the brain can easily differentiate.
The more distinct they are from one another the better.
So I'm trying to figure out ways of making them as distinct as possible.
My best attempt so far is to use HRIRs on the horizontal plane ... |
Thanks to all the people who helped me here, I finally achieved to implement a notch filter (*), with coefficients given by :
http://www-users.cs.york.ac.uk/~fisher/mkfilter/trad.html
When I do order=2 (number of poles=4, according to this website), all is working very well with :
y[n] = a0*x[n-4] + a1*x[n-3] + a2*x[n-... |
This document on HRTFs, A Spherical Far Field HRIR/HRTF Compilation, of the Neumann KU 100, talks about headphone compensation filters.
Searching Google returns http://www.ak.tu-berlin.de/fileadmin/a0135/Publikationen/2010/Brinkmann_2010_On_the_effect_of_individual_headphone_compensation_in_binaural_synthesis.pdf
These... |
I am trying to create or find a sound that localises well.
I read that emergency vehicles are changing their sirens; people found it difficult to localise the sound of the old siren (a sinusoidal sweep), and apparently this led to a lot of accidents. Apparently a single frequency is difficult to localise.
I've experime... |
So I found this answer on one of the questions. I wanted to ask how the complex part can be related with the phase shift. I wanted to ask this on the same question but it didn't allow me. Could anyone please help me understand this concept. Even if I imagine this as a spiral with the value of the imaginary part changi... |
In error correction for example for Reed-Solomon codes there are computed syndromes for received data in decoding process. What is intuitively a syndrome (I don't mean the mathematical expression for it)?
|
Related question: What are the practically relevant differences between various image resampling methods?
Bilinear and bicubic interpolation for image resampling seem to be fairly common, but biquadratic is in my experience rarely heard of. To be sure, it's available in some programs and libraries, but generally it doe... |
I have two signals, one whisper data and one normal voice data of the same sentence.
I'm trying to build a voice conversion, but just want to start off with some simple signal manipulations.
I wanted to get the excitation of several vowels first to apply the excitation to the whispered data; basically to get a Electrol... |
This question follows from Designing a sound that localises well
I am developing an experimental system for pitch training. I am giving each of the 12 pitch-classes {C CD D DE E F FG G GA A AB B } an associated consonant phoneme: {d b r ng m v sh z h l k t }
So I could sing a chromatic scale 'da ba ra nga ... etc'
My ... |
How is it possible to draw an audio frequency response graph (x-axis : freq from 20 to 20 000hz, y-axis : dB) from a simple equation like
y[n] = x[n] - x[n-486]
(Example: with this equation I get a nice comb filter that removes frequency 197,5Hz and all its harmonics, for an audio file with sampling rate 96 Khz)
How ... |
I will try to explain what is my level of understanding of this problem, please correct me if I'm wrong:
RMS is the Root Mean Square, it represent the mean value of the input signal.
PSD is the measurement of the responses that shows me at which frequencies most of the energy is concentrated.
The area below a curve is... |
In this question, we consider the Hilbert space of vectors in $\mathbb{R}^4$ with one of its basis $\{v_0, v_1, v_2, v_3\}$.
Given the first three basis vectors
$$v_0=\left[\begin{array}{c}
1/2\\
1/2\\
1/2\\
1/2
\end{array}\right]v_1=\left[\begin{array}{c}
1/2\\
1/2\\
-1/2\\
-1/2
\end{array}\right]
v_2=\left[\begin{arr... |
Does any one of you know what topic to teach people related to signal processing.
The audience is mainly for a group of software programmer working on a telecommunication product who have limited knowledge of signal processing.
Thanks!
|
I am trying to find the amplitude and phase plots of the saw tooth waveform pictured.I have already computed the Fourier series of the waveform but I don't know how to derive the amplitude and phase plots from the sawtooth's Fourier series.
$$
x(t) = \begin{cases}
\frac{V}{T}t + V & \text{$-T < t < 0$} \\... |
Currently I'm working out plotting biquads in a VST/AU plugin and have problems with the graphics part. I posted in several forums but haven't got a really useful answer for this case.
First, tell me if I'm right how to get the magnitude response. For this I use this equation
http://rs-met.com/documents/dsp/BasicDigit... |
It seems that 2D Wigner-Ville has been used to test the compression of data.
The interference is however a problem.
Can someone send me a link to a code where this has been done?
I want to study the code and change it little study the compression and Wigner distribution.
Here, seems to be basic steps to make 512x512 m... |
I've recently started playing with the Kalman filter for a simple 2D (x,y,dx,dy) tracking toy problem. But I seem to have some misunderstanding on what I can expect from the filter. I'm interested in plotting the uncertainty ellipse from the corrected covariance matrix but noticed a few observations:
The covariance de... |
For a system such that $y[n]=\cos(x[n])$, I'd like to determine whether the system is causal or not.
Definition 1: A system is causal if and only if the output $y[n]$ is a function of a linear combination of inputs $x[n-k]$ such that $k \ge0 $.
We can see that the system is causal by this definition because $y[n]$ onl... |
Wikipedia - Wiener Deconvolution shows this formula:
$$
\ G(f) = \frac{H^*(f)S(f)}{ |H(f)|^2 S(f) + N(f) }
$$
Could someone explain the derivation?
Specifically, where does the squaring ($|H(f)|^2$) come from?
|
Can anyone suggest an introduction to speech processing,
class notes or books
accompanied by programs that can be run to see what the algorithms do ?
For example, in Burrus et al., Computer-Based Exercises for Signal Processing Using Matlab,
20 of the 400 pages are on speech processing. But it's from 1994;
there must b... |
Does the entropy of a symbol represent the most optimal average number of bits that can be used to represent a symbol? For example take the example of tossing a coin:
$$
H_{source} = -p_{head} log_2 p_{head} - p_{tail} log_2 p_{tail}
$$
so if
$$
p_{head} = p_{tail} = \frac{1}{2}
$$
then H = 1 bit, but if
$$
... |
I'm new to audio (and signal processing in general), and I've been learning about the AAC codec. I've read that it supports 1-48 synchronous channels in a single stream. My question is: what need is there for 48 channels in a single audio stream? I understand that if you have a 7.1 speaker setup you can send a separate... |
FFT of some signal $x$ has two parts: $|x|$, which is the magnitude spectrum and $arg(x)$, which is the phase spectrum. I want each frequency to have the same phase. Is there a way to accomplish this?
|
I was not quite sure what the right SE for this was, so I posted this also here on SciComp. Please tell me which one to remove :)
Problem statement
I have a few hundred unrelated time series, say $P_i(t)$, with $i=1,2,...,N$. The sample times are all unequally spaced.
It is known that during certain (unknown) interva... |
I have a processing unit that does a nice EQ filtering on soundfiles.
The only thing I can do is : I give the original .wav file, and the processing unit gives me back the filtered .wav. I have no other information than this.
I'd like to know the frequency response graph of this processing unit. How to do it ?
1) feed... |
I just have a small theoretical question that buggers me to be honest.
I have a sound file that I found its spectrum on linear and on logarithmic scale. Alright, I have done this, it's a plain standard procedure. The problem is that I cant really understand the plot on logarithmic scale. (it just looks like a typical ... |
I am trying to understand how you use an FFT in matlab to measure/retrieve the magnitude of the input signal.
For my understanding I have put this script together, creating a 15Hz sine wave with a sampling frequency of 1kHz.
fftsize = 2048; % How many samples for FFT
fs = 1000; % Sampli... |
I'm trying to code the Haar wavelet approximation of a function.
I'm a complete newbie.
Edit
Below is a pseudo-code of my algorithm for $f: [0,10] \to \mathbb R$, $f(x)=x$.
I picked it here and there on google.
1) Haar wave formula of order $N$ for $x \in [0, \infty)$
$$
\psi_{N,j}(x) = \frac{1}{\sqrt{N}} \cdot\begin... |
How might one model the propagation of a sound wave through a 3D environment with obstacles in it?
My initial thought would be to use freely moving particles that repel one another; with sufficiently many particles I see no reason why the simulation wouldn't work.
However, if any particle could collide with any other t... |
I have a 3D data set sampled on a regular grid shown in the Figure below. This data can be interpreted as an image where the point altitude is equivalent to the pixel intensity value.
In the middle of my 3D data, I have a peak, that becomes a bright spot when the data is interpreted as an image.
My goal is to remove th... |
Ok, so the situation is that I have a DFII biquad with some filter coefficients:
\begin{align}
w[n] &= x[n] - a_1*w[n-1] - a_2*w[n-2]\\
y[n] &= b_0*w[n] + b_1*w[n-1] + b_2*w[n-2]
\end{align}
While the filter is running, I change the coefficients and I believe my filter is going unstable. I want to build in a check t... |
I have the image below. This image contains sets of points that are aligned horizontally and vertically.
How can i find these sets of points? based 1) on their alignment and 2) on the ** **inter-line and Inter-word distances.
Do I have to use these 2 arguments, alignment and distance? or is one of them sufficient?
I... |
I use a simple biquad IIR filter :
if (n >= 2)
y[n] = (b0/a0)*x[n] + (b1/a0)*x[n-1] + (b2/a0)*x[n-2]
- (a1/a0)*y[n-1] - (a2/a0)*y[n-2]
else
y[n] = x[n] // here it probably needs to be modified
(thanks to http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt)
It works very well.
My problem... |
This question follows from Modelling propagation of sound wave by particle simulation
This last fortnight I have started experimenting with Binaural audio; it really blows everything else out of the water.
The main reason it hasn't taken off, I think, is that every individual needs to get their own HRTF calculated, whi... |
I would like to map the envelope of a sound (which has max dB level = dbmax, and min dB level = dbmin) from [dbmin,dbmax] to [-inf,dbmax].
Here is an example :
It's like some sort of dynamic compression (or rather dynamic range 'expansion' to be precise), but here the mapping is linear from [dbmin,dbmax] to [-inf,db... |
I just want to ask, whenever you reduce a bandwidth of signal by filtering, I wonder why do you need to improve the number of bits used to represent that signal.
The number of bit growth is
$$\frac{1}{2}\log_2\left(\frac{BW_{in}}{BW_{out}}\right)$$
Is it to represent to small noise due to filtering.
Thanks!
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